Well-established power distribution systems exist throughout most of the United States, and other countries, which provide power to customers via power lines. With some modification, the infrastructure of the existing power distribution systems can be used to provide data communication in addition to power delivery, thereby forming a power line communication system (PLCS). In other words, existing power lines, that already have been run to many homes and offices, can be used to carry data signals to and from the homes and offices. These data signals are communicated on and off the power lines at various points in the power line communication system, such as, for example, near homes, offices, IP network service providers, and the like.
In one example PLCS, a backhaul point forms the gateway between the power line and conventional telecommunications medium and communicates with a plurality of downstream communication devices such as transformer bypass devices. The backhaul point and its plurality of communication devices (and their associated user devices) form a PLCS subnet. The backhaul point and its plurality of communication devices (and their associated user devices) form a PLCS subnet. In some systems, a power line repeater may be added to the power line between the backhaul point and its communications devices.
As the use of the internet becomes more and more prevalent, it has been used to transmit voice data. Initially, software was developed to allow users at two different computers connected over the internet to talk to one another using internet protocol (IP) packets. This service was often of poor quality as the voice data was subject to latency. Subsequently, interfaces between the IP network and the standard phone network were developed to allow phone calls initiated from a computer to be routed onto the public switched telephone network (PSTN) and also for calls initiated on the PSTN to be routed to a computer. Such systems have been referred to voice over IP (VoIP).
VoIP endpoints have been developed that connect directly to a high-speed IP network connection and allow for VoIP telephone service without using a computer. A VoIP endpoint is a hardware and/or software function that adapts and packetizes analog or digital telephony signals (including video telephone telephony signals) and call supervision data for transmission to and from an IP network. The IP network may include the internet or a dedicated network, such as for example, a network that provides dedicated bandwidth to a user. There a numerous examples of VoIP endpoints, for example, analog telephone adaptor (ATA), IP telephones, cordless IP telephones, WiFi (i.e., IEEE 802.11) telephone, VoIP gateways, integrated access devices (IAD), IP private branch exchanges (PBX), video telephones, softphones, or gaming consoles. The ATA typically digitizes the analog voice signal, creates voice data packets of the digitized voice data, and transmits the voice data packets to the IP network. An IP phone typically integrates an ATA and a telephone. A cordless IP phone may integrate an ATA and a cordless telephone. A WiFi phone typically integrates an ATA and a telephone and sends VoIP packets to an IP network using the WiFi standard. A VoIP gateway may convert analog voice or digital voice signals into VoIP packets. An IAD may comprise a router with analog or digital voice ports and built-in CODECs to packetize voice inputs. An IP PBX may integrate a PBX and CODECS to packetize voice. A softphone may comprise a software program that uses a microphone and speakers (or a headset) to implement a CODEC and packetization and then to transmit and receive the VoIP packets over an IP network. A gaming console may include a voice headset and integrated voice packetization and allows gamers to talk to each other over an IP network.
FIG. 1 shows an example of a prior art VoIP system. A VoIP endpoint 110 connects to a high-speed IP network interface device 120. The high-speed IP network interface device 120 connects to the IP network 130. A VoIP switch 140 may also connect to the IP network 130. A PSTN gateway 150 connects the IP network 130 to the PSTN 160. The PSTN 150 is connected to analog phones 100. In general, any two devices endpoint devices (e.g., phone 100, VoIP endpoint 110) may call and receive calls from any other device. Thus, it is possible for a first VoIP endpoint 110 to make calls to and from other VoIP endpoints 110 (either on the PLCS or other IP networks), cell phones, computer systems with interactive voice response systems, voice enhanced gaming systems, etc.
To initiate a call a user picks up a phone at the VoIP endpoint 110, and the VoIP endpoint 110 generates an analog dial tone. Note that other types of call initiation may be used in a VoIP system such as, for example, initiating a call from a network server to two users. In this example, when the user dials a number, the VoIP endpoint 110 detects the digit tones and determines the number being dialed. The VoIP endpoint 110 produces a data packet containing the phone number being dialed and sends the data packet to a VoIP switch 140. The VoIP switch 140 determines whether the phone number dialed is on the same VoIP network. If it is, the VoIP switch 140 passes the call to the IP address of the VoIP endpoint 110 associated with the phone number that was dialed. If the destination endpoint is not locally known to the VoIP switch 140, the VoIP switch 140 typically will forward the call request and call routing information to another VoIP switch or to the PSTN 160 via a PSTN gateway device 150. In this event, the PSTN 160 utilizes global network of switches and databases to route calls to any publicly registered telephone number. If the destination phone is on the PSTN 160, the PSTN gateway 150 passes the voice data packets onto the PSTN 160 for delivery to the phone called as if the call originated from another phone on the PSTN 160. The voice packets from the user receiving the call travel in the reverse direction back to the caller. Also, either the VoIP end point 110 or the VoIP switch 140 may also perform other call control operations such as call transfer, multi-party conferencing, etc.
The VoIP endpoint 110 may provide the interface between a standard analog phone and a high-speed interface device. The VoIP endpoint 110 also may provide typical telephone indication signals such as dial tone, ring tone, and busy signal. Further, when the VoIP endpoint 110 is an ATA, it may generate a ring signal to cause an analog phone to ring when a call is received. The VoIP endpoint 110 may digitize an analog voice signal using one of many standard voice CODECs. The resulting voice packets are sent to an IP address as determined as described above. Once the call is established, the VoIP endpoint 110 converts the voice packets received into an analog signal for the user.
The high-speed IP network interface device 120 connects the ATA 110 to the IP network 130. For example, in a digital subscriber line (DSL) system, the IP network interface device 120 may comprise a DSL modem, and in a cable system, the IP network interface device 120 may comprise a cable modem.
The VoIP switch 140 connected to the IP network as described above helps to set up and route phone calls. Multiple VoIP switches 140 may be used to perform load balancing of call requests, to perform different call control operations, and/or to segregate responsibility for groupings of VoIP endpoints 110. Depending on the call control protocol used, the VoIP switch 140 may receive all of the voice data traffic and route it accordingly. Otherwise, the VoIP switch 140 may provide an IP address to the VoIP endpoint 110 identifying where to send voice data packets for the call.
The PSTN gateway 150 provides an interface between the P network 130 and the PSTN 160. The PSTN gateway 150 may have an IP address so that voice packets may be sent to it. Those voice packets may then be repackaged as necessary and passed onto the PSTN 160. Also, the PSTN gateway 150 may convert the packets from one encoding scheme to another as needed. Also, voice data traffic for a user of a VoIP system that the PSTN gateway 150 receives from the PSTN 160 is then sent to an IP address so that the voice packets can be passed to the VoIP endpoint 110 corresponding to the number being called. While only one PSTN gateway 150 is shown in this example, multiple PSTN gateways 150 spread out geographically may be connected to the IP network 130.
Users of the PSTN have come to expect a high quality of service (QOS) in their phone service. Achieving a QOS that is similar to that of the standard PSTN is one of the challenges for VoIP telephone service providers. Packet latency is a significant challenge in achieving high QOS in a VoIP telephone system. If the voice packets are delayed while traversing the IP network, the delay becomes noticeable to the users. Talk over between the users results as is the case with satellite telephone links. In order to achieve a desired packet latency, a latency budget is set for the various parts of the VoIP telephone system. The IP network was designed to carry data from on point to another. The various data packets in a data stream may follow various paths between two points. If the packets arrive out of order or are delayed, they can be buffered and then reordered. Also, a lost packet may be resent. Because of the variability of data traffic on the IP network, the latency of data packets on the IP network is highly variable. For typical data traffic latency of seconds may be acceptable. This is unacceptable for voice traffic. The latency of packets traveling on the IP network 130 must be limited and is typically specified. Also, the amount of latency in the VoIP system may be managed to provide different levels of QOS for the VoIP telephone service. Additionally, to deliver high quality voice service, packet loss and jitter must also be limited.
A user of a VoIP service uses an VoIP endpoint 110 to establish and communicate over a connection. For home users, such IP network connections commonly use digital subscriber line (DSL) or DOCSIS cable connections. For business users, T1 and T3 lines have been used to provide high-speed IP network connections. Now that PLCS can deliver high-speed IP network connections, VoIP services may be provided using a PLCS system. Therefore there remains a need for a PLCS system that provides VoIP telephone service to users and that manages latency for voice traffic in order to provide different QOS levels for VoIP telephone service.